Streaming media

ABSTRACT

A method for providing a session invitation protocol (SIP) session between a first and a second entity. An SIP session is established between the first and second entity. At least one media stream is transmitted from the first entity to the second entity. At least one of transmission, storage and play back of the at least one media stream is controlled in the SIP session at the first and/or second entity.

CROSS-REFERENCE TO RELATED PATENT APPLICATIONS

This application is a Continuation of U.S. patent application Ser. No.10/355,234, filed Jan. 31, 2003, which claims priority to United KingdomPriority Application 0230301.4, filed Dec. 30, 2002, both of which areincorporated herein by reference in their entirety.

FIELD OF INVENTION

The present invention relates generally to streaming media.

BACKGROUND OF THE INVENTION

A communication system is a facility that enables communication betweentwo or more entities such as user equipment and/or other nodesassociated with the system. The communication may comprise, for example,communication of voice, data, multimedia and so on. Communication isshifting to communication networks such as IP (Internet Protocol) and/orother packet switched data networks. A communication system typicallyoperates in accordance with a given standard or specification which setsout what the various elements of the system are permitted to do and howthat should be achieved. For example, the standard or specification maydefine if the user, or more precisely, the user equipment or terminal isprovided with a circuit switched service and/or a packet switchedservice. Communication protocols and/or parameters which shall be usedfor the connection may also be defined. In other words, a specific setof “rules” on which the communication can be based needs to be definedto enable communication by means of the system. The communicationbetween the user equipment and the elements of the communication networkcan be based on an appropriate communication protocol, such as the SIP(Session initiated protocol), RTSP (real time streaming protocol) orsome other protocol.

RTSP is a standard protocol for controlling multimedia streaming onInternet. The RTSP-protocol however has some disadvantages. Inparticular, RTSP does not provide for media negotiation and the clientand server cannot dynamically select best possible media formats. Medianegotiation was handled at application level, for instance, several RTSPURL to a particular media resource could be given, each with particularmedia parameters. The RTSP DESCRIBE method can be used to retrievestream parameters from the RTSP streaming server. No negotiation is doneand the streaming is possible only if the client supports the media andcodecs listed in the response to DESCRIBE.

SIP (Session Invitation Protocol, RFC 3261) is a popular protocol usedto setup multimedia sessions. Its applications include, for instance,Internet telephony and conferencing. SIP provides means for publishingendpoint capabilities, negotiating session contents (which media areused, which codecs are used, etc.) and updating sessions. It alsoincludes means for subscribing to events and delivering event documentswhenever the event status has been changed. It can handle negotiationsfor communications sessions dynamically. However, the SIP protocol doesnot support multimedia streaming.

SUMMARY OF THE INVENTION

It is an aim of embodiments of the present invention to address theabove mentioned problems.

According to a first aspect of the present invention there is provided amethod for providing an SIP session between a first and a second entity,said method comprising the steps of establishing a SIP session betweenthe first and second entity; transmitting at least one media streambetween the first entity and the second entity and controlling at leastone of transmission, storage and play back of the at least one mediastream at said first and/or second entity.

According to a second aspect of the present invention there is provideda method of providing an SIP session between a first and a secondentity, and a RTSP session between said second entity and a thirdentity, said method comprising the steps of establishing a SIP sessionbetween the first and second entity; establishing an RTSP sessionbetween said second entity and said third mapping between SIP and RSTPinformation to allow data to pass from one of said first and thirdentities to the other of said first and third entities.

According to a third aspect of the present invention, there is provideda method of providing a SIP session between a first entity and a secondentity, said method comprising the steps of establishing a SIP sessionbetween the first and second entities; providing at least one mediastream to one of said first and second entities from the other of saidentities; and updating said at least one media stream during saidsession.

According to a fourth aspect of the present invention, there is provideda communications system comprising a first and second entity, said firstand second entity being operable to establish a SIP session between thefirst and second entity, one of said entities being arranged to transmitat least one media streams to the other entity, said system being suchthat at least one of transmission, storage and play back of the mediastreams is controlled at least one of said entities.

According to a fifth aspect of the present invention, there is provideda communications system comprising a first and second entity, said firstand second entity being operable to establish a SIP session between thefirst and second entity, said first entity being arrange to retrieve adescription of a session, said first and second entities being arrangedto establish a session therebetween, one of the entities being arrangedto send at least one media stream to the other entity and said otherentity being arranged to control the play back of the media streams.

According to a sixth aspect of the present invention, there is provideda communications system comprising a first and a second entity, saidfirst and second entity being operable to establish a SIP sessionbetween the first and second entity, one of said entities being arrangedto transmit at least one media stream to the other entity, wherein saidat least one of said media streams can be sent at one of a plurality ofdifferent playback rates.

According to a seventh aspect of the present invention, there is providea communications system comprising a first, second and third entitywherein the first and second entities are arranged to establish a SIPsession therebetween, and the second and third entities are arrange toestablish a RTSP session therebetween, said second entity being arrangedto map between SIP and RSTP information to allow data to pass from oneof said first and third entities to the other of said first and thirdentities.

According to an eighth aspect of the present invention, there isprovided a communications system comprising a first and a second entity,said first and second entity being operable to establish a SIP sessionbetween the first and second entity, one of said entities being arrangedto transmit at least one media stream to the other entity, wherein saidat least one media stream us updatable during said session.

According to a ninth aspect of the present invention, there is provideda gateway for use in a communications system, wherein said gateway isarranged to establish a SIP session with one entity and a RTSP sessionwith a further entity, said gateway being arranged to map between SIPand RSTP information to allow data to pass from one of said first andthird entities to the other of said first and third entities.

BRIEF DESCRIPTION OF THE DRAWINGS

For a better understanding of the present invention and as to how thesame may be carried into effect, reference will now be made by way ofexample to the accompanying drawings in which:

FIG. 1 illustrates streaming with SIP in which by using SIP-MEX (SIPwith media streaming extensions) streaming media content is supported inSIP domain.

FIG. 2 illustrates a gateway between a streaming client and anRTSP-server which is used to enable streaming from RTSP to SIP domain.

FIG. 3 illustrates gateway between RTSP domain and SIP media server,where the stream is fetched from the SIP Media Server to RTSP domain.

FIG. 4 illustrates an example of SIP in streaming

FIG. 5 illustrates an example of voicemail with SIP

FIG. 6 illustrates updating of stream parameters in streaming

FIG. 7 illustrates an alternative updating of stream parameters instreaming

FIGS. 8 and 9 illustrates use of gateway between RTSP and SIP, whereRTSP clients connects to a gateway and receives a video stream and RTSPmessages are mapped to SIP messages. 7

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 shows a schematic diagram of streaming with SIP, in accordancewith embodiments of the invention. A streaming client A20 is using auser application A21 to receive streaming media from a user ApplicationAll of a streaming server A10. SIP-MEX A30 is used to provide streamingfeatures to SIP.

FIG. 2 shows a schematic diagram of streaming media content between aRTSP Server B11 and a SIP User Application B21, in accordance with anembodiment of the invention. The streaming server B10 has an RTSP serverB11 which sends streaming media content via a RTSP connection B31 to aRTSP Client B41 of a gateway B40. The gateway B40 handles the mapping ofthe streaming media originating from the RTSP server B11 to a userapplication B42 of the gateway so that it is compatible with the SIP-MEXB30 connection. The user application B42 of the gateway sends thestreaming media content via the SIP-MEX B30 connection to the userapplication B21 of the streaming client B20, which is in the SIP domain.

FIG. 3 shows schematically an embodiment of the invention with a gatewayC40 between a RTSP domain C20 and a SIP media server C10, where thestream is fetched from the SIP Media Server C10 to RTSP domain C20. Theuser application C11 of the streaming server C 10 sends streaming mediacontent via a SIP-MEX connection C31 to the user application C41 of thegateway C40. The gateway C40 handles the mapping of the streaming mediaoriginating from the user application of the streaming server C11 to theRTSP server C42 so that it is compatible with the RTSP connection C30.The RTSP Server C42 sends the streaming media content via the RTSPconnection C30 to the RTSP Client C21 of the streaming client C20 whichis in the RTSP domain.

FIG. 4 illustrates messaging between a client and a server, using SIPstreaming. For the sake of clarity, features of the messages areexplained only once, and previously explained or known SIP informationis assumed to be understood by any person skilled in the art. In thisexample, the client requests streaming media, and decides that the movieor any other multimedia stream should be played. The user also decidesto stop the streaming during the movie.

The communication is started by sending a SUBSCRIBE D1 message from theclient to the server. This message is used to indicate that a clientwants to request streaming media.

The first message D1 SUBSCRIBE from Client to Server is in the followingform:

SUBSCRIBE sip:movies/matrix@media.nokia.com; user=stream SIP/2.0

From: <sip:user@nokia.com>To: <sip:movies/matrix@media.nokia.com; user=stream>

Cseq: 1 SUBSCRIBE

Call-ID: 1c5e3780-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126> Expires: 0

Event: session-descriptionAccept: application/sdp

Content-length: 0

This message is now explained in more detail:

SUBSCRIBE sip:movies/matrix@media.nokia.com; user=stream SIP/2.0

The method of the message is SUBSCRIBE. This field is used to describewhat the client is requesting and that the version of SIP used is 2.0.Use of URI (uniform resource identifier) user part “movies/matrix”defining the source of requested streaming media is additionalinformation proposed by embodiments of the invention compared to currentfeatures of SIP-protocol. Additionally, use of the URI parameter“user=stream”, which defines that streaming is to be used, is also anadditional feature proposed by embodiments of the present invention tothe current SIP-protocol. These additional features to the SIP-protocolare referred as SIP-MEX.

From: <sip:user@nokia.com>

This header field is indicating from whom the message is originating.From field is typical field in the present SIP-protocol message and thevalue of the field is in accordance with present version ofSIP-protocol. However use of SIP-MEX may require use of additionalinformation in this field, so that address of the streaming media can bedefined accurately.

To: <sip movies/matrix@media.nokia.com; user=stream>

This header field is indicating for whom the message is intended. Tofield is a typical field in the present SIP-protocol message, however inthis case the field is additionally used in embodiments of the inventionand contains additional information compared to present version of SIPidentifying the source of the multimedia stream.

Cseq: 1 SUBSCRIBE

This header field indicates the sequence number of the sent request.Cseq is a feature of the present version of SIP.

Call-ID: 1c5e3780-5834-11d6-a882-00e00307ab95

This header field indicates the identity of this connection. Call-ID isa feature of the present version of SIP.

Contact. <sip.172.21.164.126>

This header field indicates the actual SIP host-port address of theclient where the notifications from the other end will be sent. Contactis a feature of present version of SIP.

Expires: 0

This field header indicates the duration of the subscription. Theexpires value of 0 indicates that subscription is terminated immediatelyafter initial notification is sent. In other words, the sender ofSUBSCRIBE request only wants to query the current description ofstreaming resource identified by the URIsip:movies/matrix@media.nokia.com. Expires is feature of present versionof SIP.

Event: Session-Description

This header field indicates the event package to which the client issubscribing. Event names are used to separate different SIP event typesfrom each other. Event header is a feature of present version of SIP,but the session-description event type is an additional feature providedby SIP-MEX.

Accept: Application/SDP

This header field indicates that client will only accept responsescontaining the stream parameters in SDP (session description protocol)format. Accept is feature of present version of SIP.

Content-Length: 0

This header field indicates the length of the message payload (number ofcharacters). Content-length is feature of present version of SIP.

After receiving the SUBSCRIBE message D1 from the Client, the Serveranswers the Client with message 202 Accepted D2. The responseacknowledges that the other end has received the SUBSCRIBE message andthe client can start waiting for NOTIFY messages.

The message 202 Accepted from the server to the client is as follows:

SIP/2.0 202 Accepted

To: <sip:movies/matrix@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 1c5e3780-5834-11d6-a882-00e00-307ab95Contact: <sip:movies/matrix@131.228.54.140; user=stream>;Event: session-description

Cseq: 1 SUBSCRIBE Content-length: 0

SIP/2.0 202 Accepted

The status code of this response message is 202 with reason phrase“Accepted”. This field indicates the class of the response message.

Contact: <sip:movies/matrix@131.228.54.140; user=stream>

This header field indicates the source of the streaming media. In thiscase the field is only compatible with SIP-MEX because it is containsthe additional information discussed previously, as compared to presentversion of SIP.

All other fields of this message are similar to those previouslyexplained.

The server also sends to the client a NOTIFY D3 message. The serverencloses the stream parameters in the payload of the NOTIFY message andcan push the information to the client dynamically.

The message D3 NOTIFY from the server to the client:

NOTIFY sip:172.21.164.126 SIP/2.0

From: <sip:movies/matrix@media.nokia.com; user=stream>To: <sip:user@nokia.com>Call-ID: 1c5e3780-5834-11d6-a882-00e00-307ab95Contact: <sip:movies/matrix@131.228.54.140; user=stream>;

CSeq: 1 NOTIFY

Event: session-descriptionRange: smtpe-25=00:00:00-02:16:13Subscription-State: terminated; reason=timeoutContent-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 2 IN 1P4 131.228.54.140s=The Matrixu=http://us.imdb.com/Titl-e?0133093t=0 0a=control:sip:movies/matrix@media.nokia.c-om; user=streama=send onlym=audio 0 RTP/AVP 96 97 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 AMR/8000a=rtpmap:98 L16/48000/2m=video 0 RTP/AVP 26 31

The method of this message is NOTIFY, its request-URI issip:172.21.164.126. The field indicates the method name and the addresswhere the message is sent.

Range: smtpe-25=00:00:00-02:16:13

This header field indicates the length of the movie or multimediastream. Range is an additional feature provided by SIP-MEX, inembodiments of the invention.

Subscription-State: Terminated; Reason=Timeout

This header field indicates the state of the SIP event subscription. Thevalue “terminated” indicates that the subscription is no more active.The “reason” parameter indicates reason for termination. In this case,the one-shot subscription is terminated with the first notification.

Content-Type: Application/SDP

This header field indicates the MIME multimedia internet mail extensiontype of the content.

Content-length header field indicates the length of the content and is afeature of the present version of SIP.

The rest of the message belongs to its contents, which consists ofstream description in SDP language.

v=0

The version “v=” field indicates the version of SDP in use. Field v isfeature of present version of SIP.

o=media-server 1804289383 2 IN IP4 131.228.54.140

The origin “o=” field indicates the identity and version of the sessiondescription. Field o is feature of present version of SIP.

s=The Matrix

The subject “s=” field indicates the subject of the stream. Field s isfeature of present version of SIP.

u=http://usimdb.com/Title?0133093

This field indicates the URL where additional information about thesession can be obtained.

t=0 0

The time “t=” field indicates the start and stop time of the multicastsessions. In this case, the start and stop time depend on client action,and the field is included only because SDP syntax requires it. Field tis feature of the present version of SIP.

The attribute “a=” fields contain optional information about session ormedia. Field a is a feature of the present versions of SIP and SDP. TheSIP user agent will ignore the SDP attributes it does not know.

a=control:sip:movies/matrix@media.nokia.com; user=stream

The attribute “control” indicates the URI that can be used to manipulatethe media stream. The “control” attribute is only compatible withSIP-MEX because it contains additional information compared to presentversion of SIP.

a=sendonly

The send only attribute field indicates that media session is onlyunidirectional, from server towards the client. The “send only”attribute is a feature of the present version of SIP.

m=audio 0 RTP/AVP 96 97 98

The “m=” media field indicates the type of a media stream within thestreaming session. It contains the media type (audio), the transportport (0) and protocol (RTP/AVP) and then alternative media formats(expressed as RTP payload numbers). The server is able to send mediausing any of the media formats listed here. The client can choose themedia format from the list provided by the server based on its ownpreferences and the network capacity available to it. Field m is featureof present version of SIP

a=rtpmap:96 AMR-WB/16000

The “rtpmap” attribute indicates mapping from RTP payload number 96 toan audio format AMR-WB with sample rate of 16 kHz. The “rtpmap”attribute is feature of present version of SIP.

a=rtpmap:97 AMR/8000

The “rtpmap” attribute indicates mapping from RTP payload number 97 toan audio format AMR with sample rate of 8 kHz. The “rtpmap” attribute isfeature of present version of SIP.

a=rtpmap:98 L16/48000/2

The “rtpmap” attribute indicates mapping from RTP payload number 98 toan audio format L16 with sample rate of 48 kHz and two channels. The“rtpmap” attribute is feature of present version of SIP.

m=video 0 RTP/AVP 26 31

The “m=” media field indicates the media type of a stream within thesession. It contains the media type (video), the transport port (0) andprotocol (RTP/AVP) and then alternative media formats (expressed aswell-known RTP payload numbers, 26 stands for JPEG, 31 for H.261).

All other fields of this message are similar to those previouslyexplained.

After the client receives the NOTIFY (D3) message from the server, theClient sends a 200 OK message D4 to the server to confirm that it hasreceived parameters for the streaming connection.

The message 200 OK from Client to Server is as follows.

SIP/2.0 200 OK

To: <sip:user@nokia.com>From: <sip:movies/matrix@media.nokia.com; user=stream>Call-ID: 1c5e3780-5834-11d6-a882-00e00307ab95

CSeq: 1 NOTIFY Content-length: 0

The status code of this response message is 200 with reason phrase “OK”.The status code 200 indicates success.

All other fields of this message are similar to those previouslyexplained.

The user interface on the client can now show the stream information onthe user terminal. The client starts streaming the movie. The user wantsto start fast rewind with the picture starting from the end of themovie, so the Range and Scale Headers are sent in the INVITE message D5.

The message INVITE from Client to Server is as follows

INVITE sip:movies/matrix@media.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:movies/matrix@media.nokia.com; user=stream>

Cseq: 1 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126>

Range: smtpe-25=−02:16:13

Scale: −2.0

Content-type: application/sdp

Content-length: 131

v=0o=−816357754 1 IN IP4 172.21.164.126s=−t=0 0c=IN IP4 172.21.164.126a=recvonlym=audio 20582 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 20584 RTP/AVP 31

The method of this request message is INVITE with the request URIsip:movies/matrix@media.nokia.com; user=stream SIP/2.0. The fieldindicates the name of the SIP method and the SIP address of therecipient. The user part (movies/matrix) in the address specifies thestream at the server. The parameter user=stream provides the networkelements with an option to separate streaming handling from VoIP (voiceover IP) calls.

Scale: −2.0

This field indicates that the playback rate of the stream is set to 2(that is twice the normal rate) and that the direction of the stream isreversed due to the negative value. Scale is an additional featureprovided by SIP-MEX in embodiments of the present invention

c=IN IP4 172.21.164.126

The connection “c” field indicates the network address of the client.The client asks server to send the media streams to the addressspecified in field c. Field c is feature of present version of SIP.

a=recvonly

The “recvonly” attribute indicates that client wants to establish anunidirectional media session, from server towards the client. The“recvonly” attribute is a feature of the present version of SIP.

m=audio 20582 RTP/AVP 96

a=rtpmap:96 AMR-WB/16000

The media description, consisting of field m and the rtpmap attribute,indicates the type of media stream within the streaming session that theclient is prepared to receive. It contains the media type (audio), thetransport port (20582) and transport protocol (RTP/AVP) that the clientuses to receive media. After the transport protocol, the client thenlists the media format (expressed as dynamic RTP payload number on the“m=” line and the “rtpmap” attribute below it) that it expects toreceive. The server is able to use any of the media formats listed hereto send the media stream. If the streaming resource does not containmedia in the listed formats, the server can either convert the format instreaming resource to the format accepted by client, or if it cannot doconverting, decline the request. The session description in E3 containedthe formats readily supported by server, so the client should includeone of the formats listed in E3 in its request here. Field m is featureof present version of SIP.

All other fields of this message are similar to those explainedpreviously.

After receiving the INVITE message D5 from the client, the server sendsa confirmation message 200 OK D6 to the client.

The message 200 OK from the server to the client is also follows:

SIP/2.0 200 OK

To: <sip:movies/matrix@media-.nokia.com; user=stream>From: <sip:user@nokia.com>

Cseq: 1 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:movies/matrix@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

Range: smtpe-25=00:00:00-02:16:13

Scale: −1.80 Speed: 1

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=The Matrixa=control:sip:movies/matrix@131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlym=audio 32852 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 32850 RTP/AVP 26

The status code of this response message is 200 with reason phrase “OK”.The status code 200 indicates success.

Speed: 1

This field value indicates that no buffering is to be used, the serverwill send stream at the same rate as it should be played out by theclient. Speed is an additional feature provided by SIP-MEX inembodiments of the present invention.

All other fields of this message similar to those described previously.

After the client receives confirmation message 200 OK D6 from theserver, the client acknowledges the server that it has received 200 OKD6 message by sending ACK message D7 to the server.

The message ACK from the client to the server is as follows

ACK sip:movies/matrix@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:movies/matrix@media.nokia.com; user=stream>

Cseq: 1 ACK

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The method of this message is ACK.

All other fields of this message are similar to those discussed earlier.

The server starts RTP media streaming. When the client wants toterminate the stream, it sends a BYE message D8 to the server.

The message BYE from the client to the server is as follows

BYE sip:movies/matrix@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:movies/matrix@media.nokia.com; user=stream>

Cseq: 2 BYE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The method of this request message is BYE, its request-URI issip:movies/matrix@131.228.54.140; user=stream. The purpose of thismessage is to terminate the session.

All other fields of this message similar to those discussed previously.

When the server receives from the client the message BYE it sends aconfirmation message 200 OK D9 to the client and stops the streaming.

The message 200 OK from the server to the client is as follows

SIP/2.0 200 OK

To: <sip:movies/matrix@media.nokia.com; user=stream>From: <sip:user@nokia.com>

Cseq: 2 BYE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The status code of this response message is 200 with reason phrase “OK”.The status code 200 indicates success.

All other fields of this message are similar to those already described.

In a second example of using streaming media in SIP, this exampleinvolves using and controlling a voice mail box with SIP. FIG. 5illustrates how messaging between parties involved are exchanged.Individual messages and actions between the parties are explained aftereach message. For the sake of clarity, only new features in the messagesare explained. The first party, Alice, tries to call the second party,Bob, but the incoming call is redirected to an voicemail server. Alicehears a pre-recorded message from the voicemail server. Alice decides toleave a voicemail message for Bob in the voice mailbox. After recordinga voicemail message Alice wants to review the message she is leaving inthe voice mailbox. The voicemail server plays to Alice the message shehas left for Bob and then Alice tears down the connection with thevoicemail server. In this use case there are included streaming controlparameters (Range, Scale, Speed) in the SDP instead of using SIPheaders.

Communication is started by sending an INVITE message E1 from the clientto the server. This message is used to advise that a client wants torequest streaming media.

The message INVITE from Alice to Bob takes the following form:

INVITE sip:bob@nokia.com SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 1 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0o=−816357754 1 IN IP4 172.21.164.126s=−t=0 0c=IN IP4 172.21.164.126m=audio 20582 RTP/AVP 96 8 0a=rtpmap:96 AMR/8000

The fields are as already described.

When Bob receives the INVITE message E1 from Alice, the request forconnection is answered with message 302 Moved Temporarily E2

The message Moved Temporarily from Bob to Alice E2 is as follows

SIP/2.0 302 Moved Temporarily

From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 1 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:bob@mailbox.nokia.com; user=stream>Content-Type: text/html

Content-Length: 121

<html><title>Please leave a message</title><body>Please leave a message at <a href=“sip:bob@mailbox.nokia.com”>; myvoicemail</a>.</body></html>

Topic of this message is SIP/2.0 302 Moved Temporarily, which indicateseffectively that Bob is unavailable and diverts the call to the voicemail box.

In the Hypertext Markup Language (HTML) part of this message data istransferred from Bob to Alice. Alice will receive a request to leavemessage to Bob's voicemail and a hyperlink to the voice mailbox isincluded in the message. The other fields are as already described.

Alice responds to the 302 Moved Temporarily message from Bob withmessage E3-ACK to acknowledge that she have received message from Bob.

The message ACK from Alice to Bob is as follows:

ACK sip:bob@nokia.com SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 1 ACK

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-Length: 0

The fields are as already described.

Alice can now redirect her connection to the voicemail server. Alicesends an INVITE message E4 to the voicemail server.

The message INVITE form Alice to Server is as follows:

INVITE sip:bob@mailbox.nokia.com; user=stream SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 2INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0o=−816357754 1 IN IP4 172.21.164.126s=−t=0 0c=IN IP4 172.21.164.126m=audio 20582 RTP/AVP 96 8 0a=rtpmap:96 AMR/800

The topic of this message is INVITE sip:bob@mailbox.nokia.com;user=stream SIP/2.0. The URI parameter user=stream indicates thatstreaming media will be used. The header and SDP fields are as alreadydescribed.

After receiving the INVITE message from Alice, the server confirms thatit has received the message from Alice and the Server send a 200 OKmessage E5 to Alice.

The 200 OK message from the server to Alice is as follows:

SIP/2.0 200 OK

From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 2 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:bob@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140a=control:sip:bob@131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlya=range:ntp=−32.0m=audio 32852 RTP/AVP 96a=rtpmap:96 AMR/8000a=control:sip:bob@131.228.54.140; user=stream

The control attribute field indicates the URI that is used to controlthe streaming media resource. The control attribute field is onlycompatible with SiP-MEX because it contains additional informationcompared to present version of SIP, but it will be silently ignored byany SIP user-agent not supporting SIP-MEX.

a=range:ntp=−32.0

The range attribute field indicates the length of the pre-recordedmessage. The range attribute field is only compatible with SIP-MEXbecause it contains additional information compared to present versionof SIP, but it will be silently ignored by any SIP user-agent notsupporting SIP-MEX.

The other fields are as already described.

After Alice receives the 200 OK message from the server, Alice respondsto the Server with the message ACK E6 to inform the server that Alice isready to receive streaming media from the Server.

The message ACK from Alice to the Server is as follows:

ACK sip:bob@131.228.54.140; user=stream SIP/2.00From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 2 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307a-b95

Content-length: 0

The message E6 ACK is similar to message E3 ACK. In E6 ACK Alice informsthe server that streaming will be used.

After the Server receives message E6 ACK from Alice, the Server startsplaying out the pre-recorded message form Bob, which lasts 32 seconds asindicated in message 200 OK by the Server. After the pre-recordedmessage is played, Alice is given a chance to record her 60-secondvoicemail to Bob. Alice is sent an INVITE message E10 updating thesession and she is redirected to a URL specific to her message at thesame time, by using a contact header.

The message INVITE E7 sent from the server to Alice is as follows:

INVITE sip:172.21.164.126 SIP/2.0

From: <sip:bob@nokia.com>To: <sip:alice@nokia.com>

Cseq: 0 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:bob/msg.10101@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321764 IN IP4 131.228.54.140s=−t=0 0c=IN IP4 131.228.54.140a=control:sip:bob/msg.10101@131.228.54.140; user=streama=sendrecva=range:ntp=0.0-60.0m=audio 32852 RTP/AVP 96a=rtpmap:96 AMR/8000

Contact: <sip:bob/msg.10101@131.228.54.140; user=stream>

This field indicates that the streaming media connection should bedirected to the given address. In this case, the field Contact hasadditional feature provided by SIP-MEX.

a=control:sip:bobimsg.10101@131.228.54.140; user=stream

The control attribute field indicates the URI that is used to controlthe streaming media resource. The attribute is different from the URIgiven in earlier SDP, and the change indicates that the media resourceto which Alice is now being redirected is different from resource playedout earlier.

a=range:ntp=0.0-60.0

The range attribute field indicates that the length of the message Alicecan record for Bob is something between 0 seconds and 60 seconds. Therange attribute field is an additional feature, provided by embodimentsof the invention.

The other fields are as already described.

Alice then decides to leave a voicemail in to the Bob's voice mailbox.Alice sends an 200 OK message E8 to the server and confirms that shewants to leave a voicemail.

The message 200 OK from Alice to the server is as follows:

SIP/2.0 200 OK

From: <sip:bob@nokia.com>To: <sip:alice@nokia.com>

Cseq: 0 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0o=−816357754 2 IN 1P4 172.21.164.126s=−t=0 0c=IN IP4 172.21.164.126m=audio 20582 RTP/AVP 96a=rtpmap:96 AMR/8000

The fields are as already described.

After the server receives the confirmation message E8 from Alice, theserver acknowledges to Alice that it has received message E8 by sendingan acknowledgment ACK message E9 to Alice.

The message E9 from the server to Alice is as follows:

ACK sip:172.21.164.126 SIP/2.0

From: <sip:bob@nokia.com>To: <sip:alice@nokia.com>

Cseq: 0

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The fields are as already described.

Alice has now a streaming connection with the server, and Alice can nowrecord her message to Bob. However, Alice wants to review her message,which she can do it by sending an INVITE message E10 to the server. Notethat the Request URI is the URL provided in the Contact header of themessage E10 sent by the voicemail server. The same URL is also specifiedin the updated SDP. The Request URI is specific to the voicemail messagerecorded by Alice.

The message E10 from Alice to the Server is as follows

INVITE sip:bob/msg.10101@mailbox.nokia.com; user=stream SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>Cseq: 3 INVITE Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0o=−816357754 3 IN IP4 172.21.164.126s=−t=0 0c=IN IP4 172.21.164.126a=range:ntp=0.0−a=recvonlym=audio 20582 RTP/AVP 96 8 0a=rtpmap:96 AMR/8000

The fields are as already described.

After receiving message E10 from Alice, the Server answers Alice with a200 OK message E11 confirming that it has received the message fromAlice.

The message E11 from the server to Alice is as follows

SIP/2.0 200 OK

From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 3 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:bob/msg.10101@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321764 IN IP4 131.228.54.140s=−t=0 0c=IN IP4 131.228.54.140a=control:sip:bob/msg.10101@131.228.54.140; user=streama=sendonlya=range:ntp=0.0-27.6m=audio 32852 RTP/AVP 96a=rtpmap:96 AMR/8000

The fields are as already described.

After Alice receives the 200 OK message from the Server, Alice respondsto the Server with an ACK message E12 to inform the server that Alice isready to listen to the message that she has left for Bob, so that Alicereceives the streaming media from the server

The message E12 from Alice to the Server is as follows

ACK sip:bob/msg.10101@131.228.54.140; user=stream SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 3 ACK

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The fields are as already described.

After Alice has reviewed her voicemail to Bob, Alice decides todisconnect from the server, so she hangs up the call. The messageE13-BYE is sent from Alice to the server.

The message E13 BYE from Alice to the server is as follows:

BYE sip:bob/msg.10101@131.228.54.140; user=stream SIP/2.0From: <sip:alice@nokia.com>To: <sip:bob@nokia.com>

Cseq: 4 BYE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The fields are as already described.

The server receives message E13 from Alice and it responds to thatmessage by sending message E14-200 OK to Alice to confirm the teardownof the connection.

The message E14 from the server to Alice is as follows:

SIP/2.0 200 OK

From: <sip:alice@nokia.com>;To: <sip:bob@nokia.com>

Cseq: 4 BYE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0

The fields are as already described

In the third example of using streaming media in SIP, this case presentsan advanced case of use of SIP in streaming. FIG. 6 illustratesmessaging between a client and a server. In this case a client requestsstreaming a live media stream (audio) from the server. While theoriginal stream (audio) is playing, a stream (video) is added to it.Changes in the stream are made with NOTIFY message which notifies theClient about a new stream (video). The client decides to receive the newstream (video), too. The Client negotiates the media again, now withvideo support. Messaging happens simultaneously, so the original stream(audio) need not be cancelled.

Communication between the Client and Server is started by the message F1SUBSCRIBE from Client to Server. Steps from F1 to F7 are the similar tosteps D1 to D7 of FIG. 4. The full messages for messages F1 to F7 areset out below.

Message F1

SUBSCRIBE sip:reports/interim/q2-2002@-media.nokia.com; user=streamSIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@med-ia.nokia.com; user=stream>

Cseq: 1 SUBSCRIBE

Call-ID: 4df6e7d8-5834-11d6-a882-00e00307ab95

Contact: <sip:172.21.164.126> Expires: 3600

Event: session-descriptionAccept: application/sdp

Content-length: 0 Message F2 SIP/2.0 202 Accepted

To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 4df6e7d8-5834-11d6-a882--00e00307ab95

Cseq: 1 SUBSCRIBE

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Event: session-description

Content-length: 0 Message F3 NOTIFY sip:172.21.164.126 SIP/2.0

From: <sip:reports/interim-/q2-2002@media.nokia.com; user=stream>To: <sip:user@nokia.com>Call-ID: 4df6e7d8-5834-11d6-a882-00e003- 07ab95

Cseq: 1 NOTIFY

Contact: <sip:reports/interim/q2-20- 02@131.228.54.140; user=stream>Event: session-descriptionContent-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 2 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002t=0 0a=control: sip:reports/interim/q2-2002-@media.nokia.com; user=streama=sendonlym=audio 0 RTP/AVP 96 97 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 AMR/8000a=rtpmap:98 L16/48000/2

Message F4 SIP/2.0 200 OK

To: <sip:user@nokia.com>From: <sip:reports/interim/q2-20-02@media.nokia.com; user=stream>Call-ID: 4df6e7d8-5834-11d6-a882--00e00307ab95

Cseq: 1 NOTIFY Content-length: 0 Message F5

INVITE sip:reports/interim/q2-2002@media.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 3 INVITE Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0

0=−816357754 1 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonly m=audio 20582 RTP/AVP 96 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 L16/16000/2

Message F6 SIP/2.0 200 OK

To: <sip:reports/interim/q2-2002@media.nokia.com-; user=stream>From: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 4 INVITE

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002t=0 0c=IN IP4 131.228.54.140a=control: sip:reports/interim/q2-2002@media.nokia.com; user=streama=sendonlym=audio 32852 RTP/AVP 96a=rtpmap:96 AMR-WB/16000

Message F7

ACK sip:reports/interim/q2-2002@media.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-20-02@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Cseq: 4 ACK Content-length: 0

The connection between the client and the server is now established, andRTP media streaming starts. Next, the session description is updatedwithin the media server and the media server notifies the client about avideo stream that can be added to the session. The client asks server toadd the video stream to the original stream, this is done in a similarway to the way in which the original audio streaming was started. Stepsfrom F8 to F12 are similar to steps F3 to F7, so these steps are notdescribed in detail. These messages are set out below.

Message F8 NOTIFY sip: 172.21.164.126 SIP/2.0

From: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>To: <sip:user@nokia.com>Call-ID: 4df6e7d8-5834-11d6-a882-00e00307ab95

Cseq: 5 NOTIFY

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Event: session-description Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 3 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002t=0 0a=control:sip:reports/interim/q2-2002@media.nokia.com; user=streama=sendonlym=audio 0 RTP/AVP 96 97 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 AMR/8000a=rtpmap:98 L16/48000/2m=video 0 RTP/AVP 26 31

Message F9 SIP/2.0 200 OK

To: <sip:user@nokia.com>From: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 4df6e7d8-5834-11d6-a882-00e00307ab95

Cseq: 5 NOTIFY Content-length: 0 Message F10

INVITE sip:reports/interim/q2-2002@131.228.54.140:user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-20-02@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Cseq: 6 INVITE Contact: <sip:172.21.164.126>

Accept: application/sdpRange: npt=0.000−Content-type: application/sdp

Content-length: 131

v=0

0=−816357754 1 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonlym=audio 20582 RTP/AVP 96 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 L16/16000/2m=video 20584 RTP/AVP 26 31

Message F11 SIP/2.0 200 OK

To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 6 INVITE

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002a=control:sip:reports/interim/q2-2002@media.nokia.com; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlym=audio 32852 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 32850 RTP/AVP 26

Message F12

ACK sip:reports/interim/q2-2002@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 6 ACK Content-length: 0

RTP Media streaming continues now with the both audio and video streams.When the Client wants to teardown the connection, the client sendsmessage F13 BYE to the server. Steps F13 and F14 are typical connectionteardown messages and they are similar to steps D8 and D9 of FIG. 4.These messages are set out below:

Message F13

BYE sip:reports/interim/q2-2002@131.2-28.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 7 BYE Content-length: 0 Message F14 SIP/2.0 200 OK

To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Cseq: 7 BYE Content-length: 131

An alternative way to update stream parameters is shown in FIG. 7, whichillustrates messaging between a client and a server. In this exampleupdating streaming content is done by a server-initiated INVITE message.The new media stream (video) is negotiated and added simultaneouslywhile the audio stream is played.

Communication between the Client and Server is started by the message G1INVITE from Client to Server. Steps from G1 to G3 are similar to stepsE4 to E6, steps G4 to G6 are similar to steps E7 to E9 and steps G7 andG8 are similar to steps E13 and E14 of FIG. 4. The messages are set outbelow.

Message G1

INVITE sip:reports/interim/q2-2002@med-ia.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@med-ia.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00-307ab95

Cseq: 1 INVITE Contact: <sip:172.21.164.126>

Accept: application/sdp

Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0

0=−816357754 1 IN 1P4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonlym=audio 20582 RTP/AVP 96 98a=rtpmap:96 AMR-WB/16000a=rtpmap:97 L16/16000/2m=video 20584 RTP/AVP 26 31

Message G2 SIP/2.0 200 OK

To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 1 INVITE

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002t=0c=IN IP4 131.228.54.140a=control: sip:reports/interim/q2-2002@media.nokia.com; user=streama=sendonlym=audio 32852 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 0 RTP/AVP 26 31

Message G3

ACK sip:reports/interim/q2-2002@media.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 1 ACK Content-length: 0

RTP Media streaming starts with audio only.

Message G4 INVITE sip:172.21.164.126 SIP/2.0

From: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>To: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00-e00307ab95

Cseq: 1 INVITE

Contact: <sip:reports/interim/q2-2002@131.228.54.140; user=stream>Accept: application/sdpContent-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321764 IN IP4 131.228.54.140s=Nokia Interim Report Q2/2002t=0 0 c=IN IP4 131.228.54.140a=control: sip:reports/interim/q2-2002@media.nokia.com; user=streama=sendonlym=audio 33400 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 33402 RTP/AVP 26 31

Message G5 SIP/2.0 200 OK

From: <sip:reports/interim/q2-2002@media.nokia.c-om; user=stream>To: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 1 INVITE Contact: <sip:172.21.164.126>

Content-type: application/sdp

Content-length: 131

v=0

0=−816357754 2 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonlym=audio 20582 RTP/AVP 96a=rtpmap:96 AMR-WB/16000m=video 20584 RTP/AVP

Message G6 ACK sip:172.21.164.126 SIP/2.0

From: <sip:user@nokia.com>To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 1 ACK Content-length: 0

RTP Media streaming continues now with the both audio and video. Whenthe Client wants to teardown the connection, normal teardown procedureis done.

Message G7

BYE sip:reports/interim/q2-2002@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:reports/interim/q2-20-02@media.nokia.com; user=stream>Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Cseq: 3 BYE Content-length: 0 Message G8 SIP/2.0 200 OK

To: <sip:reports/interim/q2-2002@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Cseq: 3 BYE Content-length: 0

The last example describes an ordinary RTSP client connecting to agateway and it initiates receiving a video stream. FIGS. 8 and 9illustrate the messaging between RTSP client, gateway and SIP mediaserver, with FIG. 8 showing the first part of the signaling and FIG. 9showing the second part of the signaling.

The request for streaming media by the RTSP client is announced bymessage H1 DESCRIBE from the RTSP client to the gateway.

Message H1

DESCRIBE rtsp://media-gw.nokia.com/twister RTSP/1.0

CSeq: 1

This message is a RTSP message and it indicates what the RTSP client isrequesting and where from, also version of RTSP is indicated to be 1.0

When the gateway receives message H1, it maps the message so that it cancreate a SIP-MEX message from it. The gateway sends message H2-SUBSCRIBEto the SIP media server. Communication between the gateway and SIP MediaServer is similar to previous examples, so that steps from H2 to H5 aresimilar to steps D1 to D4 in FIG. 4. These messages are:

Message H2

SUBSCRIBE sip:twister@media.nokia.com;- user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>

Cseq: 1 SUBSCRIBE

Call-ID: 1c5e3780-5834-11d6-a882-00e00307ab95

Contact: <sip:131.228.237.22> Expires: 0

Event: session-descriptionAccept: application/sdp

Content-length: 0 Message H3 SIP/2.0 202 Accepted

To: <sip:twister@media.nokia.com; user=stream>From: <sip:user@nokia.com>Call-ID: 1c5e3780-5834-11d6-a882-00e003-07ab95Contact: <sip:twister@131.228.54.140; user=stream>Event: session-description

Cseq: 1 SUBSCRIBE Content-length: 0 Message H4 NOTIFY sip:131.228.237.22SIP/2.0

From: <sip:twister@media.nokia.com; user=stream>To: <sip:user@nokia.com>Call-ID: 1c5e3780-5834-11d6-a882--00e00307ab95Contact: <sip:twister@131.228.54.140; user=stream>-

CSeq: 1 NOTIFY

Event: session-description Range: smtpe-25=00:00:00-02:16:13Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 2 IN IP4 131.228.54.140s=Twistera=control:sip:twister@media.nokia-.com; user=streamt=0 0a=sendonlym=audio 0 RTP/AVP 0m=video 0 RTP/AVP 26

Message H5 SIP/2.0 200 OK

To: <sip:user@nokia.com>From: <sip:twister@media.nokia.com; user=stream>Call-ID: 1c5e3780-5834-11d6-a882-00e00307ab95

CSeq: 1 NOTIFY Content-length: 0

The gateway sends information about the requested streaming media to theRTSP client in message H6--200 OK.

Message H6 RTSP/1.0 200 OK

CSeq: 1 Content-Type: application/sdp

Content-Length: 164

v=0

0=−2890844256 2890842807 IN IP4 172.16.2.93

s=Twistera=control:rtsp://media-gw.nokia.com/twistert=0 0 m=audio 0 RTP/AVP 0 a=control:audiom=video 0 RTP/AVP 26a=control:video

Fields in this message are common to RTSP and will be clear to a personskilled in the art.

The RTSP Client now starts streaming the movie and sends messageH7-SETUP to the gateway.

Message H7

SETUP rtsp://media-gw.nokia.com/twister/audio RTSP/1.0

CSeq: 2

Transport: RTP/AVP; unicastclient_port=8000-8001

Fields in this message are common to RTSP.

The gateway receives the H7 message from RTSP client and interpretsmessage H7 to a form that the SIP media server can understand. Thegateway sends message H8-INVITE to the SIP media server. Communicationbetween the gateway and the SIP media server is similar to previousexamples, so that steps from H8 to H10 are similar to steps D5 to D7 inFIG. 4.

Message H8

INVITE sip:twister@media.nokia.com; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>Cseq: 1 INVITE Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:131.228.237.22> Content-length: 0 Message H9 SIP/2.0 200OK

To: <sip:twister@media.nokia.com; user=str-eam>From: <sip:user@nokia.com>

Cseq: 1 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:twister@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.14s=Twister a=control: sip:twister@131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlya=range:smtpe-25=00:00:00-02:16:13m=audio 9004 RTP/AVP 0a=rtpinfo: seq=92781365; rtptime=21375134m=video 9002 RTP/AVP 26a=rtpinfo: seq=9810092; rtptime=3450012

Message H10

ACK sip:twister@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; us-er=stream>

Cseq: 1 ACK

Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Content-length: 83

v=0

0=−816357754 1 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=inactivem=audio 8000 RTP/AVP 0m=video 0 RTP/AVP 26

The gateway sends confirmation to the RTSP client in message H11-200 OK.

Message H11 RTSP/1.0 200 OK CSeq: 2

Transport: RTP/AVP; unicastclient_port=8000-8001;source=131.228.54.140; server_port=9004-9005 Session: 12345678

Fields in this message are common to RTSP.

The RTSP client sets up also the video session and sends message H12SETUP to the gateway

Message H12

SETUP rtsp://media-gw.nokia.com/t-wister/video RTSP/1.0

CSeq: 3

Transport: RTP/AVP; unicast; client_port=8002-8003

Fields in this message are common to RTSP.

The Gateway confirms the request H12 for adding of the video session bysending message H13-200 OK to the RTSP client.

Message H13 RTSP/1.0 200 OK CSeq: 3

Transport: RTP/AVP; unicast; client_port=8002-8003;source=131.228.54.140; server_port=9002-9003

Session: 12345678

Fields in this message are common to RTSP.

The RTSP client requests that playing of the streaming media starts andsends message H14 to the gateway.

Message H14

PLAY rtsp://media-gw.nokia.com/twister RTSP/1.0

CSeq: 4

Range: npt=0−

Session: 12345678

The gateway requests to the SIP media server that video streaming shouldalso be included. The gateway sends message H15-INVITE to the SIP mediaserver. Communication between the gateway and the SIP media server issimilar to the previous examples, so that steps H15 to H17 are similarto steps H8 to H10.

Message H15

INVITE sip:twister@131.228.54.140-; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>

Cseq: 2 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:131.228.237.22> Content-length: 0

v=0

0=−816357754 2 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonly a=range:ntp=0−m=audio 8000 RTP/AVP 0m=video 8002 RTP/AVP 26

Message H16 SIP/2.0 200 OK

To: <sip:twister@media.nokia.com; us-er=stream>From: <sip:user@nokia.com>Cseq: 2 INVITE Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:twister@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Twistera=control: sip:twister@131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlya=range:ntp=0−m=audio 9000 RTP/AVP 0a=rtpinfo: seq=656823807; rtptime=2829945m=video 9002 RTP/AVP 26a=rtpinfo: seq=9810092; rtptime=3450012

Message H17

ACK sip:twister@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>

Cseq: 2 ACK

Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Content-length: 83

The gateway sends confirmation to the RTSP client in message H18-200 OK.

Message H18 RTSP/1.0 200 OK CSeq: 4 Session: 12345678

RTP-Info: url=rtsp://media-gw.nokia.com/t-wister/audio; seq=656823807;rtptime=2829945RTP-Info: url=rtsp://media-gw.nokia.com/twister/video; seq=9810092;rtptime=3450012

Fields in this message are common to RTSP.

The RTSP Client now receives the streaming media from the SIP MediaServer. Next the RTSP Client wants to pause the streaming media from theSIP Media Server. The

RTSP Client sends message H19-Pause to the Gateway.

Message H19

PAUSE rtsp://media-gw.nokia.com/twister RTSP/1.0

CSeq: 5 Session: 12345678

Fields in this message are common to RTSP.

Gateway receives this message H19 PAUSE and it interprets it to a formthat the SIP Media Player can understand it. Gateway sends message H20to the SIP media server to pause the streaming. Messages H20 and H21have value “inactive” in their field a of the message—this feature ofembodiments of the present invention pauses the streaming in the SIPMedia Server. Otherwise steps H20 to H22 similar to steps H8 to H10earlier in this example.

Message H20

INVITE sip:twister@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>Cseq: 3 INVITE Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:131.228.237.22> Content-length: 0

v=0

0=−816357754 2 IN 1P4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=inactivem=audio 8000 RTP/AVP 0m=video 8002 RTP/AVP

Message H21 SIP/2.0 200 OK

To: <sip:twister@media.nokia.com; user=stream>From: <sip:user@nokia.com>

Cseq: 3 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:twister@131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Twistera=control: sip:twister@131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=inactivem=audio 9000 RTP/AVP 0m=video 9002 RTP/AVP

Message H22

ACK sip:twister@131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>

Cseq: 3 ACK

Call-ID: 87ed5e72-5834-11d6-a882--00e00307ab95

Content-length: 83

RTP media is now paused. The gateway sends confirmation to the RTSPclient in message H23-200 OK.

Message H23 RTSP/1.0 200 OK CSeq: 5 Session: 12345678

Fields in this message are common to RTSP.

The RTSP client is ready to resume playing the stream, so the Clientsends message H24-PLAY to the Gateway.

Message H24 PLAY from RTSP Client to GatewayPLAY rtsp://media-gw.nokia.com/twister RTSP/1.0

CSeq: 6

Range: npt=15−

Session: 12345678

Fields in this message are common to RTSP.

The gateway requests that the SIP media server resume the streamingmedia from the point it was paused. The gateway sends message H25-INVITEto the SIP Media Server. Communication between the gateway and the SIPmedia server is similar to previous steps, that is steps H25 to H27 aresimilar to steps H13 to H15.

Message H25

INVITE sip:twister@ 131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>;To: <sip:twister@media.nokia.com; user=stream>

Cseq: 4 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Contact: <sip:131.228.237.22> Content-length: 0

v=0

0=−816357754 2 IN IP4 172.21.164.126

s=−t=0 0c=IN IP4 172.21.164.126a=recvonlya=range: npt=15−m=audio 8000 RTP/AVP 0m=video 8002 RTP/AVP 26

Message H26 SIP/2.0 200 OK

To: <sip:twister@media.nokia.com; user=stream>From: <sip:user@nokia.com>

Cseq: 4 INVITE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Contact: <sip:twister@ 131.228.54.140; user=stream>Content-type: application/sdp

Content-length: 131

v=0o=media-server 1804289383 321763 IN IP4 131.228.54.140s=Twistera=control: sip:twister@ 131.228.54.140; user=streamt=0 0c=IN IP4 131.228.54.140a=sendonlya=range: npt=15−m=audio 9000 RTP/AVP 0a=rtpinfo:seq=656823882; rtptime=2949945m=video 9002 RTP/AVP 26a=rtpinfo:seq=9810467; rtptime=3450027

Message H27

ACK sip:twister@ 131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>To: <sip:twister@media.nokia.com; user=stream>

Cseq: 4 ACK

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 83

The gateway sends confirmation to the RTSP client in message H28-200 OK.

Message H28 RTSP/1.0 200 OK CSeq: 6 Session: 12345678

RTP-Info: url=rtsp://media-gw.nokia.com/twister/audio; seq=656823882;rtptime=2949945RTP-Info: url=rtsp://media-gw.noki-a.com/twister/video; seq=9810467;rtptime=3450027

Fields in this message are common to RTSP.

The RTSP Client again receives streaming media from the SIP mediaserver. Next the RTSP client wants to stop the streaming media from theSIP media server. The RTSP Client sends message H29-TEARDOWN to theGateway.

Message H29

TEARDOWN rtsp://media-gw.nokia.com/twister RTSP/1.0

CSeq: 7 Session: 12345678

Fields in this message are common to RTSP.

The gateway receives message H29-TEARDOWN from the RTSP client. Thetypical teardown procedure is done. Message H30-BYE is sent to the SIPmedia server to start the teardown. Steps H30 and H31 are similar tosteps D8 and D9 of FIG. 4.

Message H30

BYE sip:twister@ 131.228.54.140; user=stream SIP/2.0From: <sip:user@nokia.com>;To: <sip:twister@media.nokia.com; user=stream>

Cseq: 5 BYE

Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95

Content-length: 0 Message H31 SIP/2.0 200 OK

To: <sip:twister@media.nokia.com; user=stream>From: <sip:user@nokia.com>Cseq: 5 BYE Call-ID: 87ed5e72-5834-11d6-a882-00e00307ab95Content-length: 0

The Gateway sends confirmation to the RTSP client in message H32-200 OK.

Message H32 RTSP/1.0 200 OK CSeq: 7

Fields in this message are common to RTSP.

In embodiments of the invention, a communication network is able todynamically select best format for streaming media regardless whetherthe domain of the user is based on RTSP or SIP. In embodiments of theinvention, media streaming extensions for the SIP protocol are proposedin order to create a version of SIP with media streaming extensions. Inthe embodiments, this is SIP-MEX, which adds all the RTSP functionalityto SIP. Embodiments of the invention enable mapping between RTSP andSIP-MEX, and connection between SIP and RTSP-domains can be achievedwith a gateway between them.

With SIP-MEX there is no need for a separate streaming protocol ornetwork proxies for streaming services. According to one embodiment ofthe invention, existing SIP architecture can be used to provide an easybilling scheme to streaming services, and user applications can accessstreaming media more easily. The streaming media resources, like videoclips, can be accessed with a plain Internet telephone application.

In accordance with another embodiment of the present invention, ininternet telephony SIP requests to streaming media resources can bedistinguished from telephony requests, and the streaming media can becached.

By adding the functionality of multimedia streaming SIP, it is possibleto use media streaming parameters (range, scale, speed) in the SIPmessages. In embodiments of the invention streaming media format may beselected dynamically in RTSP environment, which means updating mediasession during the streaming. On the other hand that SIP environment cansupport RTSP-functionality, which enables mapping between RTSP andSIP-MEX, and their: URLs (Uniform Resource Locator).

Embodiments of the invention provide a to establish a media session inSIP/SDP with properly negotiated medias. The SIP protocol or the SDP(Session Description Protocol) descriptions used by SIP is extended sothat they can provide all the RTSP functionality. The SIP with mediastreaming extensions (SIP-MEX) can be used in various networkconfigurations as described above.

Thus embodiments of the invention may provide a method for mappingbetween RTSP URLs and SIP URLs. The URL to the media resource is givenin RTSP request line, e.g. SETUP rtsp://media.nokia.com/movies/matrixRTSP/1.0 When a SIP element is given a media resource URL, e.g.rtsp://media.nokia.com/movies/matrix, it needs to convert the RTSP URLto a SIP URL. The mapping is straightforward: the host:port part is leftas is and the SIP username part is used to express the media resource,in this case, movies/matrix, like as follows: INVITEsip:movies/matrix@media-.nokia.com SIP/2.0 To distinguish between themedia stream resources and legacy SIP usernames, likesip:user@nokia.com, and make it possible to unambiguously map betweenSIP and RTSP URLs, the parameter user=stream can be used: INVITEsip:movies/matrix@media.nokia.com; user=stream SIP/2.0

The user=stream parameter tells the Sims registrar, proxy or applicationserver (AS) to handle the requests to media streaming resources properly(i.e., do not mix them with normal users). A proxy can also cachestreaming media accessed with a user=stream parameter. An UA processingan SIP URL with user=stream parameter can add the streaming controls(play, rewind, pause, etc.) to the user interface.

Embodiments of the invention may provides means for updating mediasession during the streaming. SIP event extension, SUBSCRIBE-NOTIFYmethods, can be used to retrieve the stream parameter data expressed inSession Description Protocol (SDP). If the stream parameters changeduring the playback, e.g. a video channel is added to a live stream, anew NOTIFY containing updated SDP can be sent to the client. The clientthen can send a new INVITE (re-INVITE) to the server and media isnegotiated again.

When using RTSP, the Range header specifies the point where the playbackstarts. No such header currently exists in SIP. There is two alternativesolutions, the Range header can be added to SIP or a Range attribute canbe added to SDP.

The syntax of the RTSP Range header is as follows:

Range=“Range”“:” 1\#ranges-specifier

[“;” “time” “=” utc-time]

ranges-specifier=npt-range|utc-range|smpte-range

Example as a SIP Header:

Range: clock=19960213T143205Z-; time=19970123T143720Z

Example as an SDP Attribute:

a=range:clock=19960213T143205Z-; time=19970123T143720Z

Scale header is used in the RTSP playback or record to view/listen themedia stream at different playback rates (implementing fast forward/fastrewind). For example, with Scale: 3.0 the server would transmit mediawith timestamps changing three times as fast as normally. Just likeRange, SIP needs to be extended with Scale header or a Scale attributecould be added to the SDP description.

Example as a SIP Header:

Scale: −1.0

Example as an SDP Attribute:

a=scale:−1.0

Speed is used to ask the server to deliver playback stream at aparticular transfer rate, for example, with Speed: 3.0 the server wouldtransmit media with playback time of 3.0 seconds within 1.0 second. Theplayback rate is not affected by transfer rate. This allows client tobuffer the stream and avoid interrupts in the playback (due to networkcongestion). Again, a new SIP header or SDP attribute can be used.

Consider the following example of a video stream. It contains 25 framesa second. The server puts a timestamp to each frame, and client playsthe frames at the time indicated by the timestamp.The stream is normally transmitted like this:

A..B..C..D..E..F..G..H..I..J..K..L..M..N..O..P..Q..R..S..T..U..V..W..X..Y..Z

And played like this

A..B..C..D..E..F..G..H..I..J..K..L..M..N..O..P..Q..R..S..T..U..V..W..X..Y..Z

In this case, the real time, the time in video clip and time indicatedby timestamps go at the same rate.With Speed: 3.0, the frames are transmitted like this:

ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789@!#$%&/©=+?®

But played out like this:A..B..C..D..E..F..G..H..I..J..K..L..M..N..O..P..Q..R..S..T..U..V..W..X..Y..ZIn this case, the time indicated by timestamps (and video time) go threetimes faster than the real time. However, the playout happens at normalrate.With Scale: 3.0, they are transmitted like this (assuming the streamingserver is intelligent enough to drop extra frames away):

A..D..G..J..M..P..S..V..Y..b..e..h..k..n..q..t..w..z..2..5..8..!..%..©..?..

And played out like this:A..D..G..J..M..P..S..V..Y..b..e..h..k..n..q..t..w..z..2..5..8..!..%..©..?..In this case, the real time and time indicated by timestamps go again atthe same rate, but the time in video clip is accelerated.

In this case, the time indicated by timestamps (and video time) go threetimes faster than the real time. However, the playout happens at normalrate. With Scale: 3.0, they are transmitted like this (assuming thestreaming server is intelligent enough to drop extra frames away):A.D.G.J.M.S.V.Y.b.e.h.k.n.q.t.w.z.2.5.8.!.%..©..?..quadrature. Andplayed out like this:A.D.G.J.M.P.S.V.Y.b.e.h.k.n.q.t.w.z.2.5.8.!.%..COPY-RGT..?..quadrature.In this case, the real time and time indicated by timestamps go again atthe same rate, but the time in video clip is accelerated.

SIP event extension, SUBSCRIBE-NOTIFY methods, can be used to retrievethe stream parameter data expressed in Session Description Protocol(SDP). If the stream parameters change during the playback, e.g., avideo channel is added to a live stream, a new NOTIFY containing updatedSDP can be sent to the client. The client then can send a new INVITE(re-iNVITE) to the server and media is negotiated again.

Parameter update can also be handled with re-INVITE, sent by thestreaming server. There is no need to use NOTIFY to bear changeinformation. The re-INVITE is now used in opposite direction, thestreaming server sends the re-INVITE and the streaming client receivesit. Note, however, that a RTSP-SIP gateway can not always honor there-INVITE sent by streaming server.

Parameter update can also be handled with re-INVITE, sent by thestreaming server. There is no need to use NOTIFY to bear changeinformation. The re-INVITE is now used in opposite direction, thestreaming server sends the re-INVITE and the streaming client receivesit.

It should be appreciated that is some embodiments of the invention, theinitially established SIP session may be without streaming with thestreaming being added and/or negotiated later. In other embodiments ofthe invention, the SIP session maybe established with a media streamsession.

The implementation is mostly done in application level. The protocollevel (SIP/SDP) implementation is by adding a support for extensionheaders/attributes. The application level implementation is done for theserver and client.

1. A method for providing an SIP session between a first and a secondentity, said method comprising the steps of: establishing a sip sessionbetween the first and second entity; transmitting at least one mediastream from the first entity to the second entity; and controlling atleast one of transmission, storage and play back of the at least onemedia stream in said sip session at said first and/or second entity. 2.A method as claimed in claim 1, wherein the playback rate of the streamis controlled
 3. A method as claimed in claim 1, wherein the transferrate of the stream is controlled.
 4. A method as claimed in claim 1,wherein the direction of the playing of the stream is controlled.
 5. Amethod as claimed in claim 1, wherein the starting point of the playingof the stream is controlled.
 6. A method as claimed in claim 1, whereinthe speed at which one of the first and second entitities transmits tothe other of the entities is controlled.
 7. A method as claimed in claim1, wherein at least one of the transmission, storage and playback iscontrolled by information provided in SIP information
 8. A method asclaimed in claim 1, wherein at least one of the transmission, (storageand playback is controlled by information provided in a SIP header
 9. Amethod as claimed in claim 1, wherein at least one of the transmission,storage and playback is controlled by information provided by an SDPattribute.
 10. A method as claimed in claim 1 comprising the furtherstep of: retrieving a description of a streaming session by the firstentity.
 11. A method as claimed in claim 10, wherein said retrievingstep uses at least one of the SIP methods SUBSCRIBE and NOTIFY
 12. Amethod as claimed in claim 1, wherein the other entity selects mediastreams it receives from a set of media streams supported by the oneentity
 13. A method as claimed in claim 10, wherein the other entityselects media streams it receives from a set of media streams supportedby the one entity.
 14. A method as claimed in claim 1, wherein the oneentity selects media streams it sends from a set of media streams thatthe other entity is prepared to receive.
 15. A method as claimed inclaim 1, wherein said one entity is one of a client and a server.
 16. Amethod as claimed in claim 1, wherein said other entity is one of aclient and a server.
 17. A method of providing an SIP session between afirst and a second entity, and a RTSP session between said second entityand a third entity, said method comprising the steps of: establishing aSIP session between the first and second entity; establishing an RTSPsession between said second entity and said third entity; mappingbetween SIP and RSTP information to allow data to pass from one of saidfirst and third entities to the other of said first and third entities.18. A method as claimed in claim 17, wherein said mapping step comprisesmapping between at least one of SIP and RSTP addresses and streamingparameters.
 19. A method as claimed in claim 17, wherein said datacomprises a media stream.
 20. A method as claimed in claim 18, whereinsaid addresses comprise URLs.